MrBesen/ffmpeg
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

simplify vorbis windowing

Browse Source 
Originally committed as revision 14205 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Loren Merritt 2008-07-13 14:56:01 +00:00
parent 6647ab80e3
commit f27e1d645e
5 changed files with 201 additions and 81 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

41
libavcodec/dsputil.c
View File

@ -3930,17 +3930,40 @@ void ff_vector_fmul_add_add_c(float *dst, const float *src0, const float *src1,
dst[i*step] = src0[i] * src1[i] + src2[i] + src3;
}
void ff_vector_fmul_window_c(float *dst, const float *src0, const float *src1, const float *win, float add_bias, int len){
int i;
for(i=0; i<len; i++)
dst[i] = src0[i]*win[len-i-1] + src1[i]*win[i] + add_bias;
}
static av_always_inline int float_to_int16_one(const float *src){
int_fast32_t tmp = *(const int32_t*)src;
if(tmp & 0xf0000){
tmp = (0x43c0ffff - tmp)>>31;
// is this faster on some gcc/cpu combinations?
// if(tmp > 0x43c0ffff) tmp = 0xFFFF;
// else tmp = 0;
}
return tmp - 0x8000;
}
void ff_float_to_int16_c(int16_t *dst, const float *src, long len){
int i;
for(i=0; i<len; i++) {
int_fast32_t tmp = ((const int32_t*)src)[i];
if(tmp & 0xf0000){
tmp = (0x43c0ffff - tmp)>>31;
// is this faster on some gcc/cpu combinations?
// if(tmp > 0x43c0ffff) tmp = 0xFFFF;
// else tmp = 0;
for(i=0; i<len; i++)
dst[i] = float_to_int16_one(src+i);
}
void ff_float_to_int16_interleave_c(int16_t *dst, const float *src, long len, int channels){
int i,j,c;
if(channels==2){
for(i=0; i<len; i++){
dst[2*i] = float_to_int16_one(src+i);
dst[2*i+1] = float_to_int16_one(src+i+len);
}
dst[i] = tmp - 0x8000;
}else{
for(c=0; c<channels; c++, src+=len)
for(i=0, j=c; i<len; i++, j+=channels)
dst[j] = float_to_int16_one(src+i);
}
}
@ -4450,7 +4473,9 @@ void dsputil_init(DSPContext* c, AVCodecContext *avctx)
c->vector_fmul = vector_fmul_c;
c->vector_fmul_reverse = vector_fmul_reverse_c;
c->vector_fmul_add_add = ff_vector_fmul_add_add_c;
c->vector_fmul_window = ff_vector_fmul_window_c;
c->float_to_int16 = ff_float_to_int16_c;
c->float_to_int16_interleave = ff_float_to_int16_interleave_c;
c->add_int16 = add_int16_c;
c->sub_int16 = sub_int16_c;
c->scalarproduct_int16 = scalarproduct_int16_c;

5
libavcodec/dsputil.h
View File

@ -63,6 +63,8 @@ void ff_h264_lowres_idct_put_c(uint8_t *dst, int stride, DCTELEM *block);
void ff_vector_fmul_add_add_c(float *dst, const float *src0, const float *src1,
const float *src2, int src3, int blocksize, int step);
void ff_vector_fmul_window_c(float *dst, const float *src0, const float *src1,
const float *win, float add_bias, int len);
void ff_float_to_int16_c(int16_t *dst, const float *src, long len);
/* encoding scans */
@ -364,10 +366,13 @@ typedef struct DSPContext {
void (*vector_fmul_reverse)(float *dst, const float *src0, const float *src1, int len);
/* assume len is a multiple of 8, and src arrays are 16-byte aligned */
void (*vector_fmul_add_add)(float *dst, const float *src0, const float *src1, const float *src2, int src3, int len, int step);
/* assume len is a multiple of 4, and arrays are 16-byte aligned */
void (*vector_fmul_window)(float *dst, const float *src0, const float *src1, const float *win, float add_bias, int len);
/* C version: convert floats from the range [384.0,386.0] to ints in [-32768,32767]
* simd versions: convert floats from [-32768.0,32767.0] without rescaling and arrays are 16byte aligned */
void (*float_to_int16)(int16_t *dst, const float *src, long len);
void (*float_to_int16_interleave)(int16_t *dst, const float *src, long len, int channels);
/* (I)DCT */
void (*fdct)(DCTELEM *block/* align 16*/);

121
libavcodec/i386/dsputil_mmx.c
View File

@ -2022,6 +2022,39 @@ static void vector_fmul_add_add_sse(float *dst, const float *src0, const float *
ff_vector_fmul_add_add_c(dst, src0, src1, src2, src3, len, step);
}
static void vector_fmul_window_sse(float *dst, const float *src0, const float *src1,
const float *win, float add_bias, int len){
#ifdef HAVE_6REGS
if(add_bias == 0){
x86_reg i = -len*2;
x86_reg j = len*2-16;
asm volatile(
"1: \n"
"movaps (%5,%0), %%xmm0 \n"
"movaps (%5,%1), %%xmm1 \n"
"movaps %%xmm0, %%xmm2 \n"
"movaps %%xmm1, %%xmm3 \n"
"shufps $0x1b, %%xmm2, %%xmm2 \n"
"shufps $0x1b, %%xmm3, %%xmm3 \n"
"mulps (%4,%0), %%xmm0 \n"
"mulps (%4,%1), %%xmm1 \n"
"mulps (%3,%0), %%xmm3 \n"
"mulps (%3,%1), %%xmm2 \n"
"addps %%xmm3, %%xmm0 \n"
"addps %%xmm2, %%xmm1 \n"
"movaps %%xmm0, (%2,%0) \n"
"movaps %%xmm1, (%2,%1) \n"
"sub $16, %1 \n"
"add $16, %0 \n"
"jl 1b \n"
:"+r"(i), "+r"(j)
:"r"(dst+len/2), "r"(src0+len/2), "r"(src1+len/2), "r"(win+len/2)
);
}else
#endif
ff_vector_fmul_window_c(dst, src0, src1, win, add_bias, len);
}
static void float_to_int16_3dnow(int16_t *dst, const float *src, long len){
// not bit-exact: pf2id uses different rounding than C and SSE
asm volatile(
@ -2083,6 +2116,87 @@ static void float_to_int16_sse2(int16_t *dst, const float *src, long len){
);
}
#define FLOAT_TO_INT16_INTERLEAVE(cpu, body) \
/* gcc pessimizes register allocation if this is in the same function as float_to_int16_interleave_sse2*/\
static av_noinline void float_to_int16_interleave2_##cpu(int16_t *dst, const float *src, long len, int channels){\
DECLARE_ALIGNED_16(int16_t, tmp[len*channels]);\
int i,j,c;\
float_to_int16_##cpu(tmp, src, len*channels);\
for(c=0; c<channels; c++){\
int16_t *ptmp = tmp+c*len;\
for(i=0, j=c; i<len; i++, j+=channels)\
dst[j] = ptmp[i];\
}\
}\
\
static void float_to_int16_interleave_##cpu(int16_t *dst, const float *src, long len, int channels){\
if(channels==1)\
float_to_int16_##cpu(dst, src, len);\
else if(channels>2)\
float_to_int16_interleave2_##cpu(dst, src, len, channels);\
else{\
float *src1;\
asm volatile(\
"shl $2, %0 \n"\
"add %0, %1 \n"\
"add %0, %2 \n"\
"lea (%2,%0), %3 \n"\
"neg %0 \n"\
body\
:"+r"(len), "+r"(dst), "+r"(src), "=r"(src1)\
);\
}\
}
FLOAT_TO_INT16_INTERLEAVE(3dnow,
"1: \n"
"pf2id (%2,%0), %%mm0 \n"
"pf2id 8(%2,%0), %%mm1 \n"
"pf2id (%3,%0), %%mm2 \n"
"pf2id 8(%3,%0), %%mm3 \n"
"packssdw %%mm1, %%mm0 \n"
"packssdw %%mm3, %%mm2 \n"
"movq %%mm0, %%mm1 \n"
"punpcklwd %%mm2, %%mm0 \n"
"punpckhwd %%mm2, %%mm1 \n"
"movq %%mm0, (%1,%0)\n"
"movq %%mm0, 8(%1,%0)\n"
"add $16, %0 \n"
"js 1b \n"
"femms \n"
)
FLOAT_TO_INT16_INTERLEAVE(sse,
"1: \n"
"cvtps2pi (%2,%0), %%mm0 \n"
"cvtps2pi 8(%2,%0), %%mm1 \n"
"cvtps2pi (%3,%0), %%mm2 \n"
"cvtps2pi 8(%3,%0), %%mm3 \n"
"packssdw %%mm1, %%mm0 \n"
"packssdw %%mm3, %%mm2 \n"
"movq %%mm0, %%mm1 \n"
"punpcklwd %%mm2, %%mm0 \n"
"punpckhwd %%mm2, %%mm1 \n"
"movq %%mm0, (%1,%0)\n"
"movq %%mm0, 8(%1,%0)\n"
"add $16, %0 \n"
"js 1b \n"
"emms \n"
)
FLOAT_TO_INT16_INTERLEAVE(sse2,
"1: \n"
"cvtps2dq (%2,%0), %%xmm0 \n"
"cvtps2dq (%3,%0), %%xmm1 \n"
"packssdw %%xmm1, %%xmm0 \n"
"movhlps %%xmm0, %%xmm1 \n"
"punpcklwd %%xmm1, %%xmm0 \n"
"movdqa %%xmm0, (%1,%0) \n"
"add $16, %0 \n"
"js 1b \n"
)
extern void ff_snow_horizontal_compose97i_sse2(IDWTELEM *b, int width);
extern void ff_snow_horizontal_compose97i_mmx(IDWTELEM *b, int width);
extern void ff_snow_vertical_compose97i_sse2(IDWTELEM *b0, IDWTELEM *b1, IDWTELEM *b2, IDWTELEM *b3, IDWTELEM *b4, IDWTELEM *b5, int width);
@ -2519,8 +2633,10 @@ void dsputil_init_mmx(DSPContext* c, AVCodecContext *avctx)
if(mm_flags & MM_3DNOW){
c->vorbis_inverse_coupling = vorbis_inverse_coupling_3dnow;
c->vector_fmul = vector_fmul_3dnow;
if(!(avctx->flags & CODEC_FLAG_BITEXACT))
if(!(avctx->flags & CODEC_FLAG_BITEXACT)){
c->float_to_int16 = float_to_int16_3dnow;
c->float_to_int16_interleave = float_to_int16_interleave_3dnow;
}
}
if(mm_flags & MM_3DNOWEXT)
c->vector_fmul_reverse = vector_fmul_reverse_3dnow2;
@ -2528,11 +2644,14 @@ void dsputil_init_mmx(DSPContext* c, AVCodecContext *avctx)
c->vorbis_inverse_coupling = vorbis_inverse_coupling_sse;
c->vector_fmul = vector_fmul_sse;
c->float_to_int16 = float_to_int16_sse;
c->float_to_int16_interleave = float_to_int16_interleave_sse;
c->vector_fmul_reverse = vector_fmul_reverse_sse;
c->vector_fmul_add_add = vector_fmul_add_add_sse;
c->vector_fmul_window = vector_fmul_window_sse;
}
if(mm_flags & MM_SSE2){
c->float_to_int16 = float_to_int16_sse2;
c->float_to_int16_interleave = float_to_int16_interleave_sse2;
}
if(mm_flags & MM_3DNOW)
c->vector_fmul_add_add = vector_fmul_add_add_3dnow; // faster than sse

111
libavcodec/vorbis_dec.c
View File

@ -149,10 +149,10 @@ typedef struct vorbis_context_s {
uint_fast8_t mode_count;
vorbis_mode *modes;
uint_fast8_t mode_number; // mode number for the current packet
uint_fast8_t previous_window;
float *channel_residues;
float *channel_floors;
float *saved;
uint_fast16_t saved_start;
float *ret;
float *buf;
float *buf_tmp;
@ -903,7 +903,7 @@ static int vorbis_parse_id_hdr(vorbis_context *vc){
vc->ret = av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
vc->buf = av_malloc( vc->blocksize[1] * sizeof(float));
vc->buf_tmp = av_malloc( vc->blocksize[1] * sizeof(float));
vc->saved_start=0;
vc->previous_window=0;
ff_mdct_init(&vc->mdct[0], bl0, 1);
ff_mdct_init(&vc->mdct[1], bl1, 1);
@ -1394,13 +1394,26 @@ void vorbis_inverse_coupling(float *mag, float *ang, int blocksize)
}
}
static void copy_normalize(float *dst, float *src, int len, int exp_bias, float add_bias)
{
int i;
if(exp_bias) {
for(i=0; i<len; i++)
((uint32_t*)dst)[i] = ((uint32_t*)src)[i] + exp_bias; // dst[k]=src[i]*(1<<bias)
} else {
for(i=0; i<len; i++)
dst[i] = src[i] + add_bias;
}
}
// Decode the audio packet using the functions above
static int vorbis_parse_audio_packet(vorbis_context *vc) {
GetBitContext *gb=&vc->gb;
uint_fast8_t previous_window=0,next_window=0;
uint_fast8_t previous_window=vc->previous_window;
uint_fast8_t mode_number;
uint_fast8_t blockflag;
uint_fast16_t blocksize;
int_fast32_t i,j;
uint_fast8_t no_residue[vc->audio_channels];
@ -1411,7 +1424,6 @@ static int vorbis_parse_audio_packet(vorbis_context *vc) {
uint_fast8_t res_chan[vc->audio_channels];
uint_fast8_t res_num=0;
int_fast16_t retlen=0;
uint_fast16_t saved_start=0;
float fadd_bias = vc->add_bias;
if (get_bits1(gb)) {
@ -1429,12 +1441,12 @@ static int vorbis_parse_audio_packet(vorbis_context *vc) {
AV_DEBUG(" Mode number: %d , mapping: %d , blocktype %d \n", mode_number, vc->modes[mode_number].mapping, vc->modes[mode_number].blockflag);
if (vc->modes[mode_number].blockflag) {
previous_window=get_bits1(gb);
next_window=get_bits1(gb);
blockflag=vc->modes[mode_number].blockflag;
blocksize=vc->blocksize[blockflag];
if (blockflag) {
skip_bits(gb, 2); // previous_window, next_window
}
blocksize=vc->blocksize[vc->modes[mode_number].blockflag];
memset(ch_res_ptr, 0, sizeof(float)*vc->audio_channels*blocksize/2); //FIXME can this be removed ?
memset(ch_floor_ptr, 0, sizeof(float)*vc->audio_channels*blocksize/2); //FIXME can this be removed ?
@ -1504,76 +1516,31 @@ static int vorbis_parse_audio_packet(vorbis_context *vc) {
// MDCT, overlap/add, save data for next overlapping FPMATH
retlen = (blocksize + vc->blocksize[previous_window])/4;
for(j=0;j<vc->audio_channels;++j) {
uint_fast8_t step=vc->audio_channels;
uint_fast16_t k;
float *saved=vc->saved+j*vc->blocksize[1]/2;
float *ret=vc->ret;
const float *lwin=vc->win[1];
const float *swin=vc->win[0];
uint_fast16_t bs0=vc->blocksize[0];
uint_fast16_t bs1=vc->blocksize[1];
float *saved=vc->saved+j*bs1/2;
float *ret=vc->ret+j*retlen;
float *buf=vc->buf;
float *buf_tmp=vc->buf_tmp;
const float *win=vc->win[blockflag&previous_window];
ch_floor_ptr=vc->channel_floors+j*blocksize/2;
vc->mdct[0].fft.imdct_calc(&vc->mdct[blockflag], buf, vc->channel_floors+j*blocksize/2, vc->buf_tmp);
saved_start=vc->saved_start;
vc->mdct[0].fft.imdct_calc(&vc->mdct[vc->modes[mode_number].blockflag], buf, ch_floor_ptr, buf_tmp);
//FIXME process channels together, to allow faster simd vector_fmul_add_add?
if (vc->modes[mode_number].blockflag) {
// -- overlap/add
if (previous_window) {
vc->dsp.vector_fmul_add_add(ret+j, buf, lwin, saved, vc->add_bias, vc->blocksize[1]/2, step);
retlen=vc->blocksize[1]/2;
} else {
int len = (vc->blocksize[1]-vc->blocksize[0])/4;
buf += len;
vc->dsp.vector_fmul_add_add(ret+j, buf, swin, saved, vc->add_bias, vc->blocksize[0]/2, step);
k = vc->blocksize[0]/2*step + j;
buf += vc->blocksize[0]/2;
if(vc->exp_bias){
for(i=0; i<len; i++, k+=step)
((uint32_t*)ret)[k] = ((uint32_t*)buf)[i] + vc->exp_bias; // ret[k]=buf[i]*(1<<bias)
} else {
for(i=0; i<len; i++, k+=step)
ret[k] = buf[i] + fadd_bias;
}
buf=vc->buf;
retlen=vc->blocksize[0]/2+len;
}
// -- save
if (next_window) {
buf += vc->blocksize[1]/2;
vc->dsp.vector_fmul_reverse(saved, buf, lwin, vc->blocksize[1]/2);
saved_start=0;
} else {
saved_start=(vc->blocksize[1]-vc->blocksize[0])/4;
buf += vc->blocksize[1]/2;
for(i=0; i<saved_start; i++)
((uint32_t*)saved)[i] = ((uint32_t*)buf)[i] + vc->exp_bias;
vc->dsp.vector_fmul_reverse(saved+saved_start, buf+saved_start, swin, vc->blocksize[0]/2);
}
if(blockflag == previous_window) {
vc->dsp.vector_fmul_window(ret, saved, buf, win, fadd_bias, blocksize/2);
} else if(blockflag > previous_window) {
vc->dsp.vector_fmul_window(ret, saved, buf+(bs1-bs0)/4, win, fadd_bias, bs0/2);
copy_normalize(ret+bs0/2, buf+(bs1+bs0)/4, (bs1-bs0)/4, vc->exp_bias, fadd_bias);
} else {
// --overlap/add
if(vc->add_bias) {
for(k=j, i=0;i<saved_start;++i, k+=step)
ret[k] = saved[i] + fadd_bias;
} else {
for(k=j, i=0;i<saved_start;++i, k+=step)
ret[k] = saved[i];
}
vc->dsp.vector_fmul_add_add(ret+k, buf, swin, saved+saved_start, vc->add_bias, vc->blocksize[0]/2, step);
retlen=saved_start+vc->blocksize[0]/2;
// -- save
buf += vc->blocksize[0]/2;
vc->dsp.vector_fmul_reverse(saved, buf, swin, vc->blocksize[0]/2);
saved_start=0;
copy_normalize(ret, saved, (bs1-bs0)/4, vc->exp_bias, fadd_bias);
vc->dsp.vector_fmul_window(ret+(bs1-bs0)/4, saved+(bs1-bs0)/4, buf, win, fadd_bias, bs0/2);
}
memcpy(saved, buf+blocksize/2, blocksize/2*sizeof(float));
}
vc->saved_start=saved_start;
return retlen*vc->audio_channels;
vc->previous_window = blockflag;
return retlen;
}
// Return the decoded audio packet through the standard api
@ -1610,8 +1577,8 @@ static int vorbis_decode_frame(AVCodecContext *avccontext,
AV_DEBUG("parsed %d bytes %d bits, returned %d samples (*ch*bits) \n", get_bits_count(gb)/8, get_bits_count(gb)%8, len);
vc->dsp.float_to_int16(data, vc->ret, len);
*data_size=len*2;
vc->dsp.float_to_int16_interleave(data, vc->ret, len, vc->audio_channels);
*data_size=len*2*vc->audio_channels;
return buf_size ;
}

4
libavutil/x86_cpu.h
View File

@ -68,6 +68,10 @@ typedef int32_t x86_reg;
# define HAVE_7REGS 1
#endif
#if defined(ARCH_X86_64) || (defined(ARCH_X86_32) && (defined(HAVE_EBX_AVAILABLE) || defined(HAVE_EBP_AVAILABLE)))
# define HAVE_6REGS 1
#endif
#if defined(ARCH_X86_64) && defined(PIC)
# define BROKEN_RELOCATIONS 1
#endif